Connected to Asterisk 1.4.17 currently running on sip (pid = 32517) <--- SIP read from 192.168.192.24:5060 ---> INVITE sip:220@ SIP/2.0 Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK1300919932;rport From: " user" ;tag=1399736409 To: Supported: replaces, 100rel, timer Call-ID: 1350920456@192.168.192.24 CSeq: 20 INVITE Session-Expires: 1800 Contact: Max-Forwards: 70 User-Agent: WirelessIP5000-v2.2.6/00:03:2a:00:53:68 Expires: 180 Content-Type: application/sdp Content-Length: 273 v=0 o=222 716452331 82168623 IN IP4 192.168.192.24 s=A_converstion c=IN IP4 192.168.192.24 t=0 0 m=audio 15000 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (14 headers 12 lines) --- Sending to 192.168.192.24 : 5060 (NAT) Using INVITE request as basis request - 1350920456@192.168.192.24 <--- Reliably Transmitting (no NAT) to 192.168.192.24:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK1300919932;received=192.168.192.24;rport=5060 From: " user" ;tag=1399736409 To: ;tag=as5e7e8131 Call-ID: 1350920456@192.168.192.24 CSeq: 20 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="domain.com", nonce="489cb9ed" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1350920456@192.168.192.24' in 32000 ms (Method: INVITE) Found user '222' sip*CLI> <--- SIP read from 192.168.192.24:5060 ---> ACK sip:220@domain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK1300919932;rport From: " user" ;tag=1399736409 To: ;tag=as5e7e8131 Call-ID: 1350920456@192.168.192.24 CSeq: 20 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- sip*CLI> <--- SIP read from 192.168.192.24:5060 ---> INVITE sip:220@domain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK168088300 From: " user" ;tag=1399736409 To: Supported: replaces, 100rel, timer Call-ID: 1350920456@192.168.192.24 CSeq: 21 INVITE Session-Expires: 1800 Contact: Proxy-Authorization: Digest username="222", realm="domain.com", nonce="489cb9ed", uri="sip:220@domain.com", response="f473af85b7023ea87cff871094bae889", algorithm=MD5 Max-Forwards: 70 User-Agent: WirelessIP5000-v2.2.6/00:03:2a:00:53:68 Expires: 180 Content-Type: application/sdp Content-Length: 273 v=0 o=222 716452331 82168623 IN IP4 192.168.192.24 s=A_converstion c=IN IP4 192.168.192.24 t=0 0 m=audio 15000 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (15 headers 12 lines) --- Sending to 192.168.192.24 : 5060 (no NAT) Using INVITE request as basis request - 1350920456@192.168.192.24 Found user '222' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.192.24:15000 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.192.24:15000 Looking for 220 in tenant-super (domain domain.com) list_route: hop: <--- Transmitting (no NAT) to 192.168.192.24:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK168088300;received=192.168.192.24 From: " user" ;tag=1399736409 To: Call-ID: 1350920456@192.168.192.24 CSeq: 21 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK6bfa50f4;rport From: ;tag=as576c7f69 To: Piano ;tag=5a4452bee2 Contact: Call-ID: 0df17c1291ce17fd CSeq: 106 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 212 confirmed --- Extension Changed 222 new state InUse for Notify User 220 -- Executing [220@tenant-super:1] GotoIf("SIP/222-b7d17278", "0?3") in new stack sip*CLI> -- Executing [220@tenant-super:2] Set("SIP/222-b7d17278", "GROUP(OUTGOING)=222") in new stack -- Executing [220@tenant-super:3] Set("SIP/222-b7d17278", "OUTBOUND_GROUP_ONCE=220@INCOMING") in new stack -- Executing [220@tenant-super:4] Set("SIP/222-b7d17278", "DB(tenant/wrapup/220/lastcall)=1199927940.15") in new stack sip*CLI> -- Executing [220@tenant-super:5] Macro("SIP/222-b7d17278", "tenant-dial|SIP/220|220|tenant|20|en||twWkKM(all-tapi^1199927940.15)||default|||vm") in new stack -- Executing [s@macro-tenant-dial:1] NoOp("SIP/222-b7d17278", ""CALL TO LOCAL EXTENSION FROM 222(WiFi)"") in new stack -- Executing [s@macro-tenant-dial:2] UserEvent("SIP/222-b7d17278", "TAPI|TAPIEVENT: LINE_NEWCALL tenant") in new stack -- Executing [s@macro-tenant-dial:3] UserEvent("SIP/222-b7d17278", "TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_OFFERING") in new stack -- Executing [s@macro-tenant-dial:4] UserEvent("SIP/222-b7d17278", "TAPI|TAPIEVENT: SET CALLERID ") in new stack sip*CLI> -- Executing [s@macro-tenant-dial:5] UserEvent("SIP/222-b7d17278", "TAPI|TAPIEVENT: LINE_CALLINFO LINECALLINFOSTATE_CALLERID") in new stack -- Executing [s@macro-tenant-dial:6] AGI("SIP/222-b7d17278", "/var/www/asterisk/telephony/scripts/agi/dial.php") in new stack -- Launched AGI Script /var/www/asterisk/telephony/scripts/agi/dial.php AGI Tx >> agi_request: /var/www/asterisk/telephony/scripts/agi/dial.php AGI Tx >> agi_channel: SIP/222-b7d17278 AGI Tx >> agi_language: en AGI Tx >> agi_type: SIP AGI Tx >> agi_uniqueid: 1199927940.15 AGI Tx >> agi_callerid: 222 AGI Tx >> agi_calleridname: WiFi AGI Tx >> agi_callingpres: 0 AGI Tx >> agi_callingani2: 0 AGI Tx >> agi_callington: 0 AGI Tx >> agi_callingtns: 0 AGI Tx >> agi_dnid: 220 AGI Tx >> agi_rdnis: unknown AGI Tx >> agi_context: macro-tenant-dial AGI Tx >> agi_extension: s AGI Tx >> agi_priority: 6 AGI Tx >> agi_enhanced: 0.0 AGI Tx >> agi_accountcode: 222 AGI Tx >> sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK6bfa50f4;rport=5060;received=192.168.192.1 From: ;tag=as576c7f69 To: Piano ;tag=5a4452bee2 Call-ID: 0df17c1291ce17fd CSeq: 106 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.192.27 sip*CLI> AGI Rx << GET VARIABLE ARG1 AGI Tx >> 200 result=1 (SIP/220) sip*CLI> AGI Rx << GET VARIABLE ARG2 AGI Tx >> 200 result=1 (220) sip*CLI> AGI Rx << GET VARIABLE ARG3 AGI Tx >> 200 result=1 (tenant) sip*CLI> AGI Rx << GET VARIABLE ARG4 AGI Tx >> 200 result=1 (20) sip*CLI> AGI Rx << GET VARIABLE ARG5 AGI Tx >> 200 result=1 (en) sip*CLI> AGI Rx << GET VARIABLE ARG6 AGI Tx >> 200 result=1 () sip*CLI> AGI Rx << GET VARIABLE ARG7 AGI Tx >> 200 result=1 (twWkKM(all-tapi^1199927940.15)) sip*CLI> AGI Rx << GET VARIABLE ARG8 AGI Tx >> 200 result=1 () sip*CLI> AGI Rx << GET VARIABLE ARG9 AGI Tx >> 200 result=1 (default) sip*CLI> AGI Rx << GET VARIABLE ARG10 AGI Tx >> 200 result=1 () sip*CLI> AGI Rx << GET VARIABLE ARG11 AGI Tx >> 200 result=1 () sip*CLI> AGI Rx << GET VARIABLE ARG12 AGI Tx >> 200 result=1 (vm) sip*CLI> AGI Rx << GET VARIABLE MACRO_PRIORITY AGI Tx >> 200 result=1 (5) sip*CLI> AGI Rx << DATABASE GET "tenant/220" "MonitorIncoming" AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE GET "tenant/220" "HotDesk" AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE GET "tenant/220" "DoNotDisturb" AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE GET "dnd" "SIP/220" AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE GET "tenant/220" "CallForward" AGI Tx >> 200 result=1 (none) sip*CLI> AGI Rx << DATABASE GET "tenant/220" "CallForwardOnBusy" AGI Tx >> 200 result=1 (none) sip*CLI> AGI Rx << DATABASE GET "tenant/220" "CallForwardOnNoAnswer" AGI Tx >> 200 result=1 (none) sip*CLI> AGI Rx << DATABASE GET "tenant/220" "CallForwardOnUnavailable" AGI Tx >> 200 result=1 (none) sip*CLI> AGI Rx << DATABASE GET "tenant/220" "CallWaiting" AGI Tx >> 200 result=0 sip*CLI> AGI Rx << GET VARIABLE INCOMINGLINE AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE GET "tenant/220" "CallForward_Internal" AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE GET "tenant/220" "CallForwardOnBusy_Internal" AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE GET "tenant/220" "CallForwardOnNoAnswer_Internal" AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE GET "tenant/220" "CallForwardOnUnavailable_Internal" AGI Tx >> 200 result=0 sip*CLI> AGI Rx << SET VARIABLE CHANNEL(language) "en" AGI Tx >> 200 result=1 sip*CLI> AGI Rx << EXEC SetMusicOnHold default -- AGI Script Executing Application: (SetMusicOnHold) Options: (default) AGI Tx >> 200 result=0 sip*CLI> AGI Rx << GET VARIABLE FORCE_RECORDING AGI Tx >> 200 result=0 sip*CLI> AGI Rx << SET VARIABLE TOUCH_MONITOR "1199927940.15" AGI Tx >> 200 result=1 sip*CLI> AGI Rx << GET VARIABLE LIMIT_IN_tenant_220 AGI Tx >> 200 result=0 sip*CLI> == Parsing '/etc/asterisk/manager.conf': Found sip*CLI> AGI Rx << VERBOSE "Extension State for '220' is '0'." 1 sip*CLI> /var/www/asterisk/telephony/scripts/agi/dial.php: Extension State for '220' is '0'. sip*CLI> AGI Tx >> 200 result=1 sip*CLI> AGI Rx << EXEC NoOp STATUS: 0 sip*CLI> -- AGI Script Executing Application: (NoOp) Options: (STATUS:) sip*CLI> AGI Tx >> 200 result=0 sip*CLI> AGI Rx << VERBOSE "Doing the action dial with params : 220" 3 sip*CLI> -- /var/www/asterisk/telephony/scripts/agi/dial.php: Doing the action dial with params : 220 sip*CLI> AGI Tx >> 200 result=1 sip*CLI> AGI Rx << DATABASE GET "tenant/222" "CallerInternalNumber" sip*CLI> AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE GET "tenant/222" "CallerInternalName" sip*CLI> AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE GET "tenant/222" "HotDesk" AGI Tx >> 200 result=0 sip*CLI> AGI Rx << DATABASE PUT "tenant/220" "CallTrace" "222" AGI Tx >> 200 result=1 sip*CLI> AGI Rx << VERBOSE "Dial string is SIP/220|20|twWkKM(all-tapi^1199927940.15)|." 1 sip*CLI> /var/www/asterisk/telephony/scripts/agi/dial.php: Dial string is SIP/220|20|twWkKM(all-tapi^1199927940.15)|. AGI Tx >> 200 result=1 sip*CLI> AGI Rx << EXEC Dial SIP/220|20|twWkKM(all-tapi^1199927940.15)| sip*CLI> -- AGI Script Executing Application: (Dial) Options: (SIP/220|20|twWkKM(all-tapi^1199927940.15)|) sip*CLI> Audio is at 192.168.192.1 port 18494 sip*CLI> Adding codec 0x4 (ulaw) to SDP sip*CLI> Adding non-codec 0x1 (telephone-event) to SDP sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.27:5060: INVITE sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK114f7624;rport From: "WiFi" ;tag=as5c2cb33f To: Contact: Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Remote-Party-ID: "WiFi" ;privacy=off;screen=no Date: Thu, 10 Jan 2008 01:19:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 242 v=0 o=root 32517 32517 IN IP4 192.168.192.1 s=session c=IN IP4 192.168.192.1 t=0 0 m=audio 18494 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 220 sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK6006f5ba;rport From: ;tag=as3af72c00 To: Piano ;tag=c4cd3ed6fa Contact: Call-ID: 4d03c8143645e9c2 CSeq: 108 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 230 early --- sip*CLI> Extension Changed 220 new state Ringing for Notify User 220 sip*CLI> Retransmitting #1 (no NAT) to 192.168.192.27:5060: INVITE sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK114f7624;rport From: "WiFi" ;tag=as5c2cb33f To: Contact: Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Remote-Party-ID: "WiFi" ;privacy=off;screen=no Date: Thu, 10 Jan 2008 01:19:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 242 v=0 o=root 32517 32517 IN IP4 192.168.192.1 s=session c=IN IP4 192.168.192.1 t=0 0 m=audio 18494 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- sip*CLI> Retransmitting #1 (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK6006f5ba;rport From: ;tag=as3af72c00 To: Piano ;tag=c4cd3ed6fa Contact: Call-ID: 4d03c8143645e9c2 CSeq: 108 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 230 early --- sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK6006f5ba;rport=5060;received=192.168.192.1 From: ;tag=as3af72c00 To: Piano ;tag=c4cd3ed6fa Call-ID: 4d03c8143645e9c2 CSeq: 108 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.192.27 sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK114f7624;rport=5060;received=192.168.192.1 From: "WiFi" ;tag=as5c2cb33f To: Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 CSeq: 102 INVITE Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK114f7624;rport=5060;received=192.168.192.1 From: "WiFi" ;tag=as5c2cb33f To: ;tag=1882511989 Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Call-Info: ;appearance-index=1 Contact: Piano Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- sip*CLI> -- SIP/220-08462d90 is ringing <--- Transmitting (no NAT) to 192.168.192.24:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK168088300;received=192.168.192.24 From: " user" ;tag=1399736409 To: ;tag=as5dbfe5cd Call-ID: 1350920456@192.168.192.24 CSeq: 21 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK114f7624;rport=5060;received=192.168.192.1 From: "WiFi" ;tag=as5c2cb33f To: ;tag=1882511989 Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Call-Info: ;appearance-index=1 Contact: Piano Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- sip*CLI> -- SIP/220-08462d90 is ringing sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK6006f5ba;rport=5060;received=192.168.192.1 From: ;tag=as3af72c00 To: Piano ;tag=c4cd3ed6fa Call-ID: 4d03c8143645e9c2 CSeq: 108 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.192.27 sip*CLI> == Parsing '/etc/asterisk/manager.conf': Found sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK114f7624;rport=5060;received=192.168.192.1 From: "WiFi" ;tag=as5c2cb33f To: ;tag=1882511989 Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Call-Info: ;appearance-index=1 Contact: Piano Server: Aastra 57i/2.1.2.30 Session-Expires: 90;refresher=uas Supported: timer, replaces Content-Type: application/sdp Content-Length: 237 v=0 o=MxSIP 0 0 IN IP4 192.168.192.27 s=SIP Call c=IN IP4 192.168.192.27 t=0 0 m=audio 3000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (15 headers 12 lines) --- sip*CLI> Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.192.27:3000 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.192.27:3000 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.192.27, port 5060 Transmitting (no NAT) to 192.168.192.27:5060: ACK sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK485a87c4;rport From: "WiFi" ;tag=as5c2cb33f To: ;tag=1882511989 Contact: Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 CSeq: 102 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Remote-Party-ID: "WiFi" ;privacy=off;screen=no Content-Length: 0 --- sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK504ef355;rport From: ;tag=as3af72c00 To: Piano ;tag=c4cd3ed6fa Contact: Call-ID: 4d03c8143645e9c2 CSeq: 109 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 212 confirmed --- Extension Changed 220 new state InUse for Notify User 220 -- SIP/220-08462d90 answered SIP/222-b7d17278 -- Executing [s@macro-all-tapi:1] UserEvent("SIP/220-08462d90", "TAPI|TAPIEVENT [~1199927940.15] LINE_CALLSTATE LINECALLSTATE_CONNECTED") in new stack Audio is at 192.168.192.1 port 17290 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.192.24:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK168088300;received=192.168.192.24 From: " user" ;tag=1399736409 To: ;tag=as5dbfe5cd Call-ID: 1350920456@192.168.192.24 CSeq: 21 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 32517 32517 IN IP4 192.168.192.1 s=session c=IN IP4 192.168.192.1 t=0 0 m=audio 17290 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> sip*CLI> <--- SIP read from 192.168.192.24:5060 ---> ACK sip:220@192.168.192.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK1601459428 From: " user" ;tag=1399736409 To: ;tag=as5dbfe5cd Call-ID: 1350920456@192.168.192.24 CSeq: 21 ACK Max-Forwards: 70 User-Agent: WirelessIP5000-v2.2.6/00:03:2a:00:53:68 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK504ef355;rport=5060;received=192.168.192.1 From: ;tag=as3af72c00 To: Piano ;tag=c4cd3ed6fa Call-ID: 4d03c8143645e9c2 CSeq: 109 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.192.27 sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> INVITE sip:222@192.168.192.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bKcbba8059c485515d0 Max-Forwards: 70 From: ;tag=1882511989 To: "WiFi" ;tag=as5c2cb33f Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 CSeq: 23354 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Piano Session-Expires: 90 Supported: timer, 100rel, replaces User-Agent: Aastra 57i/2.1.2.30 Content-Type: application/sdp Content-Length: 599 v=0 o=MxSIP 0 1 IN IP4 192.168.192.27 s=SIP Call c=IN IP4 192.168.192.27 t=0 0 m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a sip*CLI> =rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:30 a=sendonly <-------------> --- (15 headers 25 lines) --- Sending to 192.168.192.27 : 5060 (no NAT) Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 106 Found RTP audio format 107 Found RTP audio format 113 Found RTP audio format 110 Found RTP audio format 111 Found RTP audio format 112 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 115 Found RTP audio format 96 Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.192.27:3000 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found unknown media description format BV16 for ID 106 Found unknown media description format BV32 for ID 107 Found audio description format L16 for ID 113 Found audio description format PCMU for ID 110 Found audio description format PCMA for ID 111 Found audio description format L16 for ID 112 Found unknown media description format G726-16 for ID 98 Found unknown media description format G726-24 for ID 97 Found audio description format G726-32 for ID 115 Found unknown media description format G726-40 for ID 96 Found audio description format G722 for ID 9 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x4 (ulaw), peer - audio=0x194c (ulaw|alaw|g726|slin|g729|g722)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.192.27:3000 <--- Transmitting (no NAT) to 192.168.192.27:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bKcbba8059c485515d0;received=192.168.192.27 From: ;tag=1882511989 To: "WiFi" ;tag=as5c2cb33f Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 CSeq: 23354 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 192.168.192.1 port 18494 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bKcbba8059c485515d0;received=192.168.192.27 From: ;tag=1882511989 To: "WiFi" ;tag=as5c2cb33f Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 CSeq: 23354 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 32517 32518 IN IP4 192.168.192.1 s=session c=IN IP4 192.168.192.1 t=0 0 m=audio 18494 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> -- Started music on hold, class 'default', on channel 'SIP/222-b7d17278' Reliably Transmitting (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK45cf8a09;rport From: ;tag=as3af72c00 To: Piano ;tag=c4cd3ed6fa Contact: Call-ID: 4d03c8143645e9c2 CSeq: 110 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 328 confirmed --- Extension Changed 220 new state Hold for Notify User 220 sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK45cf8a09;rport=5060;received=192.168.192.1 From: ;tag=as3af72c00 To: Piano ;tag=c4cd3ed6fa Call-ID: 4d03c8143645e9c2 CSeq: 110 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.192.27 sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> ACK sip:222@192.168.192.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bKe55464be86d560b0b Max-Forwards: 70 From: ;tag=1882511989 To: "WiFi" ;tag=as5c2cb33f Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 CSeq: 23354 ACK User-Agent: Aastra 57i/2.1.2.30 Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> INVITE sip:700@domain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bKabcbbcdbec17b65f0 Max-Forwards: 70 From: Piano ;tag=076ed97773 To: 700 Call-ID: 123f597e8466bdd3 CSeq: 31087 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Piano Session-Expires: 90 Supported: timer, 100rel, replaces User-Agent: Aastra 57i/2.1.2.30 Content-Type: application/sdp Content-Length: 599 v=0 o=MxSIP 0 0 IN IP4 192.168.192.27 s=SIP Call c=IN IP4 192.168.192.27 t=0 0 m=audio 3002 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (15 headers 25 lines) --- Sending to 192.168.192.27 : 5060 (no NAT) Using INVITE request as basis request - 123f597e8466bdd3 <--- Reliably Transmitting (no NAT) to 192.168.192.27:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bKabcbbcdbec17b65f0;received=192.168.192.27 From: Piano ;tag=076ed97773 To: 700 ;tag=as0c07b10d Call-ID: 123f597e8466bdd3 CSeq: 31087 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="domain.com", nonce="4e2b3741" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '123f597e8466bdd3' in 32000 ms (Method: INVITE) Found user '220' sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> ACK sip:700@domain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bKabcbbcdbec17b65f0 Max-Forwards: 70 From: Piano ;tag=076ed97773 To: 700 ;tag=as0c07b10d Call-ID: 123f597e8466bdd3 CSeq: 31087 ACK User-Agent: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> INVITE sip:700@domain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bKf6fe578b2afe57806 Proxy-Authorization: Digest username="220",realm="domain.com",nonce="4e2b3741",uri="sip:700@domain.com:5060",response="48b1e50a9d91c70ef47b4eafc28ae1da",algorithm=MD5 Max-Forwards: 70 From: Piano ;tag=076ed97773 To: 700 Call-ID: 123f597e8466bdd3 CSeq: 31088 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Piano Session-Expires: 90 Supported: timer, 100rel, replaces User-Agent: Aastra 57i/2.1.2.30 Content-Type: application/sdp Content-Length: 599 v=0 o=MxSIP 0 0 IN IP4 192.168.192.27 s=SIP Call c=IN IP4 192.168.192.27 t=0 0 m=audio 3002 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 sip*CLI> a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (16 headers 25 lines) --- Sending to 192.168.192.27 : 5060 (no NAT) Using INVITE request as basis request - 123f597e8466bdd3 Found user '220' Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 106 Found RTP audio format 107 Found RTP audio format 113 Found RTP audio format 110 Found RTP audio format 111 Found RTP audio format 112 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 115 Found RTP audio format 96 Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.192.27:3002 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found unknown media description format BV16 for ID 106 Found unknown media description format BV32 for ID 107 Found audio description format L16 for ID 113 Found audio description format PCMU for ID 110 Found audio description format PCMA for ID 111 Found audio description format L16 for ID 112 Found unknown media description format G726-16 for ID 98 Found unknown media description format G726-24 for ID 97 Found audio description format G726-32 for ID 115 Found unknown media description format G726-40 for ID 96 Found audio description format G722 for ID 9 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x4 (ulaw), peer - audio=0x194c (ulaw|alaw|g726|slin|g729|g722)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.192.27:3002 Looking for 700 in tenant-super (domain domain.com) list_route: hop: <--- Transmitting (no NAT) to 192.168.192.27:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bKf6fe578b2afe57806;received=192.168.192.27 From: Piano ;tag=076ed97773 To: 700 Call-ID: 123f597e8466bdd3 CSeq: 31088 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> sip*CLI> -- Executing [700@tenant-super:1] Park("SIP/220-b7d200b8", "") in new stack sip*CLI> Audio is at 192.168.192.1 port 18966 sip*CLI> Adding codec 0x4 (ulaw) to SDP sip*CLI> Adding non-codec 0x1 (telephone-event) to SDP sip*CLI> <--- Reliably Transmitting (no NAT) to 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bKf6fe578b2afe57806;received=192.168.192.27 From: Piano ;tag=076ed97773 To: 700 ;tag=as581239f4 Call-ID: 123f597e8466bdd3 CSeq: 31088 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 32517 32517 IN IP4 192.168.192.1 s=session c=IN IP4 192.168.192.1 t=0 0 m=audio 18966 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> ACK sip:700@192.168.192.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bKd44c2f4be0040573c Proxy-Authorization: Digest username="220",realm="domain.com",nonce="4e2b3741",uri="sip:700@192.168.192.1",response="526796fdc15de8b67642337cdac2d0ae",algorithm=MD5 Max-Forwards: 70 From: Piano ;tag=076ed97773 To: 700 ;tag=as581239f4 Call-ID: 123f597e8466bdd3 CSeq: 31088 ACK User-Agent: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- sip*CLI> -- Started music on hold, class 'default', on channel 'SIP/220-b7d200b8' == Parked SIP/220-b7d200b8 on 701@parkedcalls. Will timeout back to extension [tenant-super] s, 1 in 45 seconds -- Added extension '701' priority 1 to parkedcalls == Spawn extension (tenant-super, s, 1) exited KEEPALIVE on 'SIP/220-b7d200b8' sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.28:5060: NOTIFY sip:225@192.168.192.28;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK7df504f1;rport From: 701 ;tag=as6607ef7f To: Office ;tag=84488dc3a2e453b Contact: Call-ID: 82c5c33cb44183351703e5cc9d880266@192.168.192.28 CSeq: 106 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 212 confirmed --- sip*CLI> Extension Changed 701 new state InUse for Notify User 225 sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK74255087;rport From: ;tag=as65992175 To: Piano ;tag=e0bb65d4b2 Contact: Call-ID: daf16b0add687e93 CSeq: 106 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 212 confirmed --- sip*CLI> Extension Changed 701 new state InUse for Notify User 220 Reliably Transmitting (no NAT) to 192.168.192.19:2063: NOTIFY sip:233@192.168.192.19:2063;line=rngpzv8j SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK11adfb4d;rport From: ;tag=as110bbcee To: ;tag=ro3em5aa41 Contact: Call-ID: 3c46f7f88191-ecjjxkove334 CSeq: 107 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 207 confirmed --- Extension Changed 701 new state InUse for Notify User 233 Reliably Transmitting (no NAT) to 192.168.192.29:5060: NOTIFY sip:221@192.168.192.29;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK6028260f;rport From: 701 ;tag=as73df5844 To: Basement ;tag=da4aa935d99053c Contact: Call-ID: ed70d93060070cfb5792d4cb3bbe7c19@192.168.192.29 CSeq: 106 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 212 confirmed --- Extension Changed 701 new state InUse for Notify User 221 sip*CLI> <--- SIP read from 192.168.192.28:5060 ---> SIP/2.0 400 Out Of Order Call-ID: 82c5c33cb44183351703e5cc9d880266@192.168.192.28 CSeq: 106 NOTIFY From: 701 ;tag=as6607ef7f To: Office ;tag=84488dc3a2e453b Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK7df504f1;rport Content-Length: 0 User-Agent: Aastra 480i/1.4.2.3000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 400 "Out Of Order" back from 192.168.192.28 sip*CLI> <--- SIP read from 192.168.192.29:5060 ---> SIP/2.0 400 Out Of Order Call-ID: ed70d93060070cfb5792d4cb3bbe7c19@192.168.192.29 CSeq: 106 NOTIFY From: 701 ;tag=as73df5844 To: Basement ;tag=da4aa935d99053c Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK6028260f;rport Content-Length: 0 User-Agent: Aastra 480i/1.4.2.3000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 --- sip*CLI> <--- SIP read from 192.168.192.19:2063 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK11adfb4d;rport=5060 From: ;tag=as110bbcee To: ;tag=ro3em5aa41 Call-ID: 3c46f7f88191-ecjjxkove334 CSeq: 107 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK74255087;rport=5060;received=192.168.192.1 From: ;tag=as65992175 To: Piano ;tag=e0bb65d4b2 Call-ID: daf16b0add687e93 CSeq: 106 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.192.27 sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> REFER sip:222@192.168.192.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bK4f3de81c5a329bc90 Max-Forwards: 70 From: ;tag=1882511989 To: "WiFi" ;tag=as5c2cb33f Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 CSeq: 23355 REFER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Piano Refer-To: 700 Referred-By: Supported: timer User-Agent: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Call 674d08511fb4eba55519d7962a583e52@192.168.192.1 got a SIP call transfer from caller: (REFER)! SIP transfer to extension 700@tenant-super by 220@domain.com <--- Transmitting (no NAT) to 192.168.192.27:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bK4f3de81c5a329bc90;received=192.168.192.27 From: ;tag=1882511989 To: "WiFi" ;tag=as5c2cb33f Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 CSeq: 23355 REFER User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Stopped music on hold on SIP/220-b7d200b8 -- Stopped music on hold on SIP/222-b7d17278 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.192.27, port 5060 Reliably Transmitting (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK69e4ba03;rport From: "WiFi" ;tag=as5c2cb33f To: ;tag=1882511989 Contact: Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 CSeq: 103 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Remote-Party-ID: "WiFi" ;privacy=off;screen=no Event: refer;id=23355 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 16 SIP/2.0 200 OK --- sip*CLI> Scheduling destruction of SIP dialog '123f597e8466bdd3' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.192.27, port 5060 Reliably Transmitting (no NAT) to 192.168.192.27:5060: BYE sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK319c8e97;rport From: 700 ;tag=as581239f4 To: Piano ;tag=076ed97773 Call-ID: 123f597e8466bdd3 CSeq: 102 BYE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 --- -- Started music on hold, class 'default', on channel 'SIP/222-b7d17278' sip*CLI> Scheduling destruction of SIP dialog '674d08511fb4eba55519d7962a583e52@192.168.192.1' in 6400 ms (Method: REFER) AGI Tx >> 200 result=-1 == Spawn extension (macro-tenant-dial, s, 6) exited non-zero on 'SIP/220-b7d200b8' -- Executing [h@macro-tenant-dial:1] ResetCDR("SIP/220-b7d200b8", "w") in new stack sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK331f3104;rport From: ;tag=as3af72c00 To: Piano ;tag=c4cd3ed6fa Contact: Call-ID: 4d03c8143645e9c2 CSeq: 111 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 213 terminated --- sip*CLI> Extension Changed 220 new state Idle for Notify User 220 sip*CLI> -- Executing [h@macro-tenant-dial:2] NoCDR("SIP/220-b7d200b8", "") in new stack sip*CLI> -- Executing [h@macro-tenant-dial:3] UserEvent("SIP/220-b7d200b8", "TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_IDLE") in new stack -- Executing [h@macro-tenant-dial:4] System("SIP/220-b7d200b8", "/var/www/asterisk/telephony/scripts/billing/cdr.sh 1199927940.15") in new stack --- sip*CLI> Retransmitting #1 (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK69e4ba03;rport From: "WiFi" ;tag=as5c2cb33f To: ;tag=1882511989 Contact: Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 CSeq: 103 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Remote-Party-ID: "WiFi" ;privacy=off;screen=no Event: refer;id=23355 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 16 SIP/2.0 200 OK --- sip*CLI> Retransmitting #1 (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK331f3104;rport From: ;tag=as3af72c00 To: Piano ;tag=c4cd3ed6fa Contact: Call-ID: 4d03c8143645e9c2 CSeq: 111 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 213 terminated --- sip*CLI> Retransmitting #2 (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK69e4ba03;rport From: "WiFi" ;tag=as5c2cb33f To: ;tag=1882511989 Contact: Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 CSeq: 103 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Remote-Party-ID: "WiFi" ;privacy=off;screen=no Event: refer;id=23355 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 16 SIP/2.0 200 OK --- sip*CLI> Retransmitting #2 (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK331f3104;rport From: ;tag=as3af72c00 To: Piano ;tag=c4cd3ed6fa Contact: Call-ID: 4d03c8143645e9c2 CSeq: 111 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 213 terminated --- sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK69e4ba03;rport=5060;received=192.168.192.1 From: "WiFi" ;tag=as5c2cb33f To: ;tag=1882511989 Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 CSeq: 103 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK319c8e97;rport=5060;received=192.168.192.1 From: 700 ;tag=as581239f4 To: Piano ;tag=076ed97773 Call-ID: 123f597e8466bdd3 CSeq: 102 BYE Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK331f3104;rport=5060;received=192.168.192.1 From: ;tag=as3af72c00 To: Piano ;tag=c4cd3ed6fa Call-ID: 4d03c8143645e9c2 CSeq: 111 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.192.27 Really destroying SIP dialog '123f597e8466bdd3' Method: ACK sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK69e4ba03;rport=5060;received=192.168.192.1 From: "WiFi" ;tag=as5c2cb33f To: ;tag=1882511989 Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 CSeq: 103 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK331f3104;rport=5060;received=192.168.192.1 From: ;tag=as3af72c00 To: Piano ;tag=c4cd3ed6fa Call-ID: 4d03c8143645e9c2 CSeq: 111 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.192.27 sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK69e4ba03;rport=5060;received=192.168.192.1 From: "WiFi" ;tag=as5c2cb33f To: ;tag=1882511989 Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 CSeq: 103 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK331f3104;rport=5060;received=192.168.192.1 From: ;tag=as3af72c00 To: Piano ;tag=c4cd3ed6fa Call-ID: 4d03c8143645e9c2 CSeq: 111 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 sip*CLI> <--- SIP read from 192.168.192.24:5060 ---> BYE sip:220@192.168.192.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK2063657786 From: " user" ;tag=1399736409 To: ;tag=as5dbfe5cd Call-ID: 1350920456@192.168.192.24 CSeq: 22 BYE Max-Forwards: 70 User-Agent: WirelessIP5000-v2.2.6/00:03:2a:00:53:68 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 192.168.192.24 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.192.24:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK2063657786;received=192.168.192.24 From: " user" ;tag=1399736409 To: ;tag=as5dbfe5cd Call-ID: 1350920456@192.168.192.24 CSeq: 22 BYE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> sip*CLI> -- Stopped music on hold on SIP/222-b7d17278 == SIP/222-b7d17278 got tired of being parked sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK3f6af02e;rport From: ;tag=as576c7f69 To: Piano ;tag=5a4452bee2 Contact: Call-ID: 0df17c1291ce17fd CSeq: 107 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 213 terminated --- Extension Changed 222 new state Idle for Notify User 220 Reliably Transmitting (no NAT) to 192.168.192.28:5060: NOTIFY sip:225@192.168.192.28;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK26cf87c4;rport From: 701 ;tag=as6607ef7f To: Office ;tag=84488dc3a2e453b Contact: Call-ID: 82c5c33cb44183351703e5cc9d880266@192.168.192.28 CSeq: 107 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 213 terminated --- Extension Changed 701 new state Idle for Notify User 225 Reliably Transmitting (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK2a5ea5cd;rport From: ;tag=as65992175 To: Piano ;tag=e0bb65d4b2 Contact: Call-ID: daf16b0add687e93 CSeq: 107 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 213 terminated --- Extension Changed 701 new state Idle for Notify User 220 Reliably Transmitting (no NAT) to 192.168.192.19:2063: NOTIFY sip:233@192.168.192.19:2063;line=rngpzv8j SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK2b0553c4;rport From: ;tag=as110bbcee To: ;tag=ro3em5aa41 Contact: Call-ID: 3c46f7f88191-ecjjxkove334 CSeq: 108 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 208 terminated --- Extension Changed 701 new state Idle for Notify User 233 Reliably Transmitting (no NAT) to 192.168.192.29:5060: NOTIFY sip:221@192.168.192.29;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK28ce2f81;rport From: 701 ;tag=as73df5844 To: Basement ;tag=da4aa935d99053c Contact: Call-ID: ed70d93060070cfb5792d4cb3bbe7c19@192.168.192.29 CSeq: 107 NOTIFY User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 213 terminated --- Extension Changed 701 new state Idle for Notify User 221 Really destroying SIP dialog '1350920456@192.168.192.24' Method: BYE sip*CLI> <--- SIP read from 192.168.192.29:5060 ---> SIP/2.0 400 Out Of Order Call-ID: ed70d93060070cfb5792d4cb3bbe7c19@192.168.192.29 CSeq: 107 NOTIFY From: 701 ;tag=as73df5844 To: Basement ;tag=da4aa935d99053c Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK28ce2f81;rport Content-Length: 0 User-Agent: Aastra 480i/1.4.2.3000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 400 "Out Of Order" back from 192.168.192.29 sip*CLI> <--- SIP read from 192.168.192.28:5060 ---> SIP/2.0 400 Out Of Order Call-ID: 82c5c33cb44183351703e5cc9d880266@192.168.192.28 CSeq: 107 NOTIFY From: 701 ;tag=as6607ef7f To: Office ;tag=84488dc3a2e453b Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK26cf87c4;rport Content-Length: 0 User-Agent: Aastra 480i/1.4.2.3000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 400 "Out Of Order" back from 192.168.192.28 sip*CLI> <--- SIP read from 192.168.192.19:2063 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK2b0553c4;rport=5060 From: ;tag=as110bbcee To: ;tag=ro3em5aa41 Call-ID: 3c46f7f88191-ecjjxkove334 CSeq: 108 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK3f6af02e;rport=5060;received=192.168.192.1 From: ;tag=as576c7f69 To: Piano ;tag=5a4452bee2 Call-ID: 0df17c1291ce17fd CSeq: 107 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.192.27 sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK2a5ea5cd;rport=5060;received=192.168.192.1 From: ;tag=as65992175 To: Piano ;tag=e0bb65d4b2 Call-ID: daf16b0add687e93 CSeq: 107 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.192.27 sip*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.192.27, port 5060 Reliably Transmitting (no NAT) to 192.168.192.27:5060: BYE sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK2dff315f;rport From: "WiFi" ;tag=as5c2cb33f To: ;tag=1882511989 Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 sip*CLI> CSeq: 104 BYE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Remote-Party-ID: "WiFi" ;privacy=off;screen=no Content-Length: 0 --- Scheduling destruction of SIP dialog '674d08511fb4eba55519d7962a583e52@192.168.192.1' in 6400 ms (Method: REFER) sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK2dff315f;rport=5060;received=192.168.192.1 From: "WiFi" ;tag=as5c2cb33f To: ;tag=1882511989 Call-ID: 674d08511fb4eba55519d7962a583e52@192.168.192.1 CSeq: 104 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Server: Aastra 57i/2.1.2.30 Content-Length: 0 --- sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.19:2063: OPTIONS sip:233@192.168.192.19:2063;line=rngpzv8j SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK31a802f1;rport From: "asterisk" ;tag=as51dc4421 To: Contact: Call-ID: 06f90faa600f018a105708791f406f74@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:19:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.19:2063 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK31a802f1;rport=5060 From: "asterisk" ;tag=as51dc4421 To: Call-ID: 06f90faa600f018a105708791f406f74@192.168.192.1 CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom320/7.1.30 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Content-Length: 0 --- sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.19:2063: OPTIONS sip:133@192.168.192.19:2063;line=h8l8tsqw SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK2dae7e2a;rport From: "asterisk" ;tag=as6daf9672 To: Contact: Call-ID: 22ea3d5839ca36d10d3391c677d6bc0d@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:19:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.19:2063 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK2dae7e2a;rport=5060 From: "asterisk" ;tag=as6daf9672 To: Call-ID: 22ea3d5839ca36d10d3391c677d6bc0d@192.168.192.1 CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom320/7.1.30 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.28:5060 ---> SIP/2.0 200 OK Call-ID: 7bf08fdf5371ab5504ee60246bdac027@192.168.192.1 CSeq: 102 OPTIONS From: "asterisk" ;tag=as23ada9fa To: ;tag=fe26c3fe529d0e1 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK74eafe72;rport Content-Length: 0 Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: Supported: replaces User-Agent: Aastra 480i/1.4.2.3000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (11 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '7bf08fdf5371ab5504ee60246bdac027@192.168.192.1' Method: OPTIONS sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.29:5060: OPTIONS sip:221@192.168.192.29;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK2ce89f4b;rport From: "asterisk" ;tag=as0f8eae51 To: Contact: Call-ID: 4da4c0731344d61e2181a5b72c4e2ccc@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:19:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.29:5060 ---> SIP/2.0 200 OK Call-ID: 4da4c0731344d61e2181a5b72c4e2ccc@192.168.192.1 CSeq: 102 OPTIONS From: "asterisk" ;tag=as0f8eae51 To: ;tag=9772675f2c306ea Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK2ce89f4b;rport Content-Length: 0 Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: Supported: replaces User-Agent: Aastra 480i/1.4.2.3000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (11 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '4da4c0731344d61e2181a5b72c4e2ccc@192.168.192.1' Method: OPTIONS sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.27:5060: OPTIONS sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK480bd5f9;rport From: "asterisk" ;tag=as4920b3ef To: Contact: Call-ID: 5af67f611f120abd4ffb6a28504c0835@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:19:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK480bd5f9;rport=5060;received=192.168.192.1 From: "asterisk" ;tag=as4920b3ef To: ;tag=568869475 Call-ID: 5af67f611f120abd4ffb6a28504c0835@192.168.192.1 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '5af67f611f120abd4ffb6a28504c0835@192.168.192.1' Method: OPTIONS sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.28:5060: OPTIONS sip:125@192.168.192.28;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK7a1ea8ab;rport From: "asterisk" ;tag=as79c65c19 To: Contact: Call-ID: 216a20f436e0229276ae6282488aa6a4@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:19:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.28:5060 ---> SIP/2.0 200 OK Call-ID: 216a20f436e0229276ae6282488aa6a4@192.168.192.1 CSeq: 102 OPTIONS From: "asterisk" ;tag=as79c65c19 To: ;tag=b148813653c5695 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK7a1ea8ab;rport Content-Length: 0 Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: Supported: replaces User-Agent: Aastra 480i/1.4.2.3000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (11 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '216a20f436e0229276ae6282488aa6a4@192.168.192.1' Method: OPTIONS sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.29:5060: OPTIONS sip:121@192.168.192.29;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK0cb67b66;rport From: "asterisk" ;tag=as08bf2972 To: Contact: Call-ID: 2cb933683b1f706a28eb0122169a9794@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:19:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.29:5060 ---> SIP/2.0 200 OK Call-ID: 2cb933683b1f706a28eb0122169a9794@192.168.192.1 CSeq: 102 OPTIONS From: "asterisk" ;tag=as08bf2972 To: ;tag=e2dbd136612986e Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK0cb67b66;rport Content-Length: 0 Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: Supported: replaces User-Agent: Aastra 480i/1.4.2.3000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (11 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '2cb933683b1f706a28eb0122169a9794@192.168.192.1' Method: OPTIONS sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.27:5060: OPTIONS sip:120@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK22e8d3fd;rport From: "asterisk" ;tag=as6d8c8b38 To: Contact: Call-ID: 401c396f0082280160d568ed048312e7@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:19:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK22e8d3fd;rport=5060;received=192.168.192.1 From: "asterisk" ;tag=as6d8c8b38 To: ;tag=1412969712 Call-ID: 401c396f0082280160d568ed048312e7@192.168.192.1 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '401c396f0082280160d568ed048312e7@192.168.192.1' Method: OPTIONS sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.24:5060: OPTIONS sip:222@192.168.192.24 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5a8059eb;rport From: "asterisk" ;tag=as4b683ac5 To: Contact: Call-ID: 5e0d77d472c7367255ac2c51653adaa2@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Thu, 10 Jan 2008 01:19:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.24:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5a8059eb;rport=5060 From: "asterisk" ;tag=as4b683ac5 To: ;tag=1594990977 Call-ID: 5e0d77d472c7367255ac2c51653adaa2@192.168.192.1 CSeq: 102 OPTIONS Accept: application/sdp Content-Length: 0 <-------------> --- (8 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '5e0d77d472c7367255ac2c51653adaa2@192.168.192.1' Method: OPTIONS